Hi Kaleem, thank you for your reply. I solved my problem with miniupnpd. I actually hoped to do it without any firewall rules. I thought when I register SIP with my gateway provider that freeswitch would initiate the audio via rtp over UDP and I don’t have not to change anything. But my provider always starts the audio. So it seems to be there is no way around some firewall rules even though it is just during the call. Whats nice though is that I only open ports for RTP during the call by limiting the port range for miniupnpd. SIP is keep open by freeswitch without any forwarding. Odd for me is that the RTCP data is not accepted from my provider. Freeswitch is not opening any port via upnpd for it. Regards, Paul 2018-07-10 15:58 GMT+02:00 kaleem rehman <k4kaleem@gmail.com>:
Hi Paul,
its likely to be firewall blocking ports.
looking at the log entry { *m=audio 19394 RTP/AVP 8 100* }
other party is using RTP port : 19394 , looking at port, its likely they are using RTP range of 16384 to 32767, is this port allowed on your side? as this is the standard RTP range.
regards, Kaleem
---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Wed, 4 Jul 2018 18:48:50 +0200 Subject: Re: [Freeswitch-users] 481 Call does not Exit Solved it.
2018-07-03 22:27 GMT+02:00 Paul Muaddib <paul.muaddib83@gmail.com>:
Hi,
if I call someone, who accepts the call and then hangs up, the call does not end on my side
Error message: 481 Call does not Exit
What is the reason for this?
Best regards, Paul
---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 00:10:01 +0200 Subject: [Freeswitch-users] Inbound call one way audio, outbound call works Hi,
when I get inbound calls I can here the caller but the caller can't here the callee Other way around, making outbound calls is not a problem. Caller and callee can hear each other
(My Setup) Phone -> FS -> NAT -> Gateway
<param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/>
What's the problem?
Name Type Data State ============================================================ ===================================== 10.0.200.2 alias internal ALIASED external profile sip:mod_ sofia@10.0.200.2:5080 RUNNING (0) external::sip-trunk.telekom.de gateway sip:XXXXXXXXXXXX@ sip-trunk.telekom.de REGED internal profile sip:mod_ sofia@10.0.200.2:5060 RUNNING (0) ============================================================ =====================================
sofia status profile internal
============================================================ ===================================== Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.200.2,10.0.200.2 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5060 BIND-URL sip:mod_sofia@10.0.200.2:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@10.0.200.2:5066;transport=ws WSS-BIND-URL sips:mod_sofia@10.0.200.2:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE info CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 6 FAILED-CALLS-IN 1 CALLS-OUT 6 FAILED-CALLS-OUT 4 REGISTRATIONS 7
sofia status profile external
============================================================ ===================================== Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5080 BIND-URL sip:mod_sofia@10.0.200.2:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 3 FAILED-CALLS-IN 1 CALLS-OUT 2 FAILED-CALLS-OUT 0 REGISTRATIONS 0
nat_map status
Nat Type: NAT-PMP, ExtIP: 87.157.68.118 NAT port mapping enabled.
I only map RTP ports. SIP is registered via TCP
Console debug level 7
2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7091 Duplicate SDP v=0 o=- 0 2 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20
2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4767 sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de. 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:24.925380 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:24.945382 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [ready][200] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7094 Remote SDP: v=0 o=- 0 3 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20
2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4767 sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de. 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:31.625959 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:31.665962 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [ready][200]
I am trying for days now but google is no real help.
Regards, Paul
---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 14:53:46 +0200 Subject: [Freeswitch-users] LOSE_RACE, 3436 Originate Failed. Cause: NONE Hi,
sometimes I have the rare case that I lose a call. I can't reproduce it. It seems a bit random. When this happens all group members have a LOSE_RACE and the caller gets disconnected? Why?
Thanks for helping
Regards, Paul
<extension name="Operator"> <condition field="destination_number" expression="^(operator)$|^(90)$" require-nested="false"> <action application="export" data="dialed_extension=operator"/>
<action application="set" data="dialed_user=$1@${domain_name}"/> <action application="set" data="call_timeout=90"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/>
<action application="bind_digit_action" data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/> <action application="digit_action_set_realm" data="get_digits"/>
<action application="ring_ready"/>
<condition field="${office_status}" expression="^(open)$" break="on-true"> <action application="bridge" data="group/buero :_: pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition> <condition field="${office_status}" expression="^(closed)$" break="on-true"> <action application="bridge" data="group/buero :_: group/werkstatt,pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition>
</condition> </extension>
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